Kristan Posted October 15, 2007 Report Share Posted October 15, 2007 Hi all, I have a strange problem with an installation of PBXnSIP. The customer is complaining of audio cutting out on some calls, and sure enough they are right. I started a wireshark capture and luckily enough managed to get a conversation with a break up almost straight away. The setup is: ISDN-> Parlay ISDN Gateway --- SIP/RTP ---> PBXnSIP --- SIP/RTP----> phones From the streams, the breakup appears to be due to the PBX not sending RTP traffic. At the time of the breakup, between the parlay and the PBX I can see a steady stream of RTP going TO the PBX, but nothing FROM it. At the same time, between the phone and the PBX, I can see traffic FROM the phone, but nothing TO it. It's like the PBX just decides to stop sending RTP, then carries on again. The duration of the break is about 8 seconds. There was also another call in progress, same as above, but terminating at our SIP trunk instead of the ISDN gateway, which also sufference idential loss for exactly the same period of time. It doesn't appear to be network related, as I can see both sides of each stream arriving at the PBX, they just don't get passed through, it's like the PBX just stops for 8 seconds, then starts again. Is there anything that would cause this? Quote Link to comment Share on other sites More sharing options...
Bill H Posted October 15, 2007 Report Share Posted October 15, 2007 Hi all, I have a strange problem with an installation of PBXnSIP. The customer is complaining of audio cutting out on some calls, and sure enough they are right. I started a wireshark capture and luckily enough managed to get a conversation with a break up almost straight away. The setup is: ISDN-> Parlay ISDN Gateway --- SIP/RTP ---> PBXnSIP --- SIP/RTP----> phones From the streams, the breakup appears to be due to the PBX not sending RTP traffic. At the time of the breakup, between the parlay and the PBX I can see a steady stream of RTP going TO the PBX, but nothing FROM it. At the same time, between the phone and the PBX, I can see traffic FROM the phone, but nothing TO it. It's like the PBX just decides to stop sending RTP, then carries on again. The duration of the break is about 8 seconds. There was also another call in progress, same as above, but terminating at our SIP trunk instead of the ISDN gateway, which also sufference idential loss for exactly the same period of time. It doesn't appear to be network related, as I can see both sides of each stream arriving at the PBX, they just don't get passed through, it's like the PBX just stops for 8 seconds, then starts again. Is there anything that would cause this? Quote Link to comment Share on other sites More sharing options...
Bill H Posted October 15, 2007 Report Share Posted October 15, 2007 Hi all, I have a strange problem with an installation of PBXnSIP. The customer is complaining of audio cutting out on some calls, and sure enough they are right. I started a wireshark capture and luckily enough managed to get a conversation with a break up almost straight away. The setup is: ISDN-> Parlay ISDN Gateway --- SIP/RTP ---> PBXnSIP --- SIP/RTP----> phones From the streams, the breakup appears to be due to the PBX not sending RTP traffic. At the time of the breakup, between the parlay and the PBX I can see a steady stream of RTP going TO the PBX, but nothing FROM it. At the same time, between the phone and the PBX, I can see traffic FROM the phone, but nothing TO it. It's like the PBX just decides to stop sending RTP, then carries on again. The duration of the break is about 8 seconds. There was also another call in progress, same as above, but terminating at our SIP trunk instead of the ISDN gateway, which also sufference idential loss for exactly the same period of time. It doesn't appear to be network related, as I can see both sides of each stream arriving at the PBX, they just don't get passed through, it's like the PBX just stops for 8 seconds, then starts again. Is there anything that would cause this? We had a similar experience. But not as long an interval of missing voice. We determined that our problem (at least at this point) was due to a high number of BLF assignments on Aastra phones. (Notify) We trimmed down the number of BLF's and the trouble went away. We are adding a few each day and watching and listening for the trouble to return or not. We have 10 to 15 very active callers on the phones all day long and if there is a problem we hear about it very quickly. Hope this can help. Bill H Quote Link to comment Share on other sites More sharing options...
Kristan Posted October 15, 2007 Author Report Share Posted October 15, 2007 We trimmed down the number of BLF's and the trouble went away. We are adding a few each day and watching and listening for the trouble to return or not.Bill H Thanks Bill, we had something along the same lines at another client with Snom 360's, thought in that instance it was the phones that were having a fit as they were receiving lots of notifies all at once as a hunt group rang. Seems we have all the difficult clients In this case it's definitely the server, as I can see the RTP from the phone to the PBX is still ok. I have a suspicion it may be OS/hardware/something other software related, but I'd like to know what it would take to cause the PBX to stop forwarding RTP. I wonder is it possible to have the PBXnSIP service run at "above normal" or "high" priority in case it's something on the server hogging the CPU or blocking. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 16, 2007 Report Share Posted October 16, 2007 We heared about cases where heavy BLF load caused a lot of jitter, but that should not stop RTP! The PBX writes a log message when it stops RTP dues to one-way audio. The 2.1 contains a new settins "timeout_hold", maybe you can try to set it to a value like 3600 (seconds) and see if the behavior changes. Quote Link to comment Share on other sites More sharing options...
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